1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
|
/*
* Opus parser for Ogg
* Copyright (c) 2012 Nicolas George
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <string.h>
#include "libavutil/intreadwrite.h"
#include "avformat.h"
#include "internal.h"
#include "oggdec.h"
struct oggopus_private {
int need_comments;
unsigned pre_skip;
int64_t cur_dts;
};
#define OPUS_SEEK_PREROLL_MS 80
#define OPUS_HEAD_SIZE 19
static int opus_header(AVFormatContext *avf, int idx)
{
struct ogg *ogg = avf->priv_data;
struct ogg_stream *os = &ogg->streams[idx];
AVStream *st = avf->streams[idx];
struct oggopus_private *priv = os->private;
uint8_t *packet = os->buf + os->pstart;
if (!priv) {
priv = os->private = av_mallocz(sizeof(*priv));
if (!priv)
return AVERROR(ENOMEM);
}
if (os->flags & OGG_FLAG_BOS) {
if (os->psize < OPUS_HEAD_SIZE || (AV_RL8(packet + 8) & 0xF0) != 0)
return AVERROR_INVALIDDATA;
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = AV_CODEC_ID_OPUS;
st->codec->channels = AV_RL8 (packet + 9);
priv->pre_skip = AV_RL16(packet + 10);
st->codec->delay = priv->pre_skip;
/*orig_sample_rate = AV_RL32(packet + 12);*/
/*gain = AV_RL16(packet + 16);*/
/*channel_map = AV_RL8 (packet + 18);*/
if (ff_alloc_extradata(st->codec, os->psize))
return AVERROR(ENOMEM);
memcpy(st->codec->extradata, packet, os->psize);
st->codec->sample_rate = 48000;
av_codec_set_seek_preroll(st->codec,
av_rescale(OPUS_SEEK_PREROLL_MS,
st->codec->sample_rate, 1000));
avpriv_set_pts_info(st, 64, 1, 48000);
priv->need_comments = 1;
return 1;
}
if (priv->need_comments) {
if (os->psize < 8 || memcmp(packet, "OpusTags", 8))
return AVERROR_INVALIDDATA;
ff_vorbis_stream_comment(avf, st, packet + 8, os->psize - 8);
priv->need_comments--;
return 1;
}
return 0;
}
static int opus_duration(uint8_t *src, int size)
{
unsigned nb_frames = 1;
unsigned toc = src[0];
unsigned toc_config = toc >> 3;
unsigned toc_count = toc & 3;
unsigned frame_size = toc_config < 12 ? FFMAX(480, 960 * (toc_config & 3)) :
toc_config < 16 ? 480 << (toc_config & 1) :
120 << (toc_config & 3);
if (toc_count == 3) {
if (size<2)
return AVERROR_INVALIDDATA;
nb_frames = src[1] & 0x3F;
} else if (toc_count) {
nb_frames = 2;
}
return frame_size * nb_frames;
}
static int opus_packet(AVFormatContext *avf, int idx)
{
struct ogg *ogg = avf->priv_data;
struct ogg_stream *os = &ogg->streams[idx];
AVStream *st = avf->streams[idx];
struct oggopus_private *priv = os->private;
uint8_t *packet = os->buf + os->pstart;
int ret;
if (!os->psize)
return AVERROR_INVALIDDATA;
if ((!os->lastpts || os->lastpts == AV_NOPTS_VALUE) && !(os->flags & OGG_FLAG_EOS)) {
int seg, d;
int duration;
uint8_t *last_pkt = os->buf + os->pstart;
uint8_t *next_pkt = last_pkt;
duration = 0;
seg = os->segp;
d = opus_duration(last_pkt, os->psize);
if (d < 0) {
os->pflags |= AV_PKT_FLAG_CORRUPT;
return 0;
}
duration += d;
last_pkt = next_pkt = next_pkt + os->psize;
for (; seg < os->nsegs; seg++) {
next_pkt += os->segments[seg];
if (os->segments[seg] < 255 && next_pkt != last_pkt) {
int d = opus_duration(last_pkt, next_pkt - last_pkt);
if (d > 0)
duration += d;
last_pkt = next_pkt;
}
}
os->lastpts =
os->lastdts = os->granule - duration;
}
if ((ret = opus_duration(packet, os->psize)) < 0)
return ret;
os->pduration = ret;
if (os->lastpts != AV_NOPTS_VALUE) {
if (st->start_time == AV_NOPTS_VALUE)
st->start_time = os->lastpts;
priv->cur_dts = os->lastdts = os->lastpts -= priv->pre_skip;
}
priv->cur_dts += os->pduration;
if ((os->flags & OGG_FLAG_EOS)) {
int64_t skip = priv->cur_dts - os->granule + priv->pre_skip;
skip = FFMIN(skip, os->pduration);
if (skip > 0) {
os->pduration = skip < os->pduration ? os->pduration - skip : 1;
os->end_trimming = skip;
av_log(avf, AV_LOG_DEBUG,
"Last packet was truncated to %d due to end trimming.\n",
os->pduration);
}
}
return 0;
}
const struct ogg_codec ff_opus_codec = {
.name = "Opus",
.magic = "OpusHead",
.magicsize = 8,
.header = opus_header,
.packet = opus_packet,
.nb_header = 1,
};
|
